| c74b93fdf3ddb6fdab7878edab4d304fb14917f9 |
|
02-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Keep effects sessions active when the caller dies.
Don't remove effects until the session they are in goes away or all
AudioEffects have been explicitly released. This allows the control
panel process to die without stopping the effects.
Change-Id: I4496e5df080230ca1af149dec95c1309ab8ea888
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 678cc9590325cb30b4bede6f2a43fe6c25bae20b |
|
27-Jul-2011 |
Eric Laurent <elaurent@google.com> |
AudioRecord: Fix getInput()
AudioRecord::getInput() was issuing a query to get a new input stream from
audio policy service instead of returning the cached input stream in AudioRecord.
Change-Id: Ice324b7c60bc0898149023797bcb56a72091b9d3
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 464d5b3da21c84ba13dc69c611d40f6bed49badb |
|
18-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Audio framework: support for audio pre processing
Audio effect framework is extended to suport effects on
output and input audio path.
AudioFlinger: Support for audio effects and effect chains is
moved from PlaybackThread class to ThreadBase class so that
RecordThread can manage effects.
Effects of type pre processing are allowed on record thread
only. When a pre processing is enabled, the effect interface handle is
passed down to the input stream so that the audio HAL can call the
process function. The record thread loop calls the effect chain process
function that will only manage the effect state and commands and skip the
process function.
AudioRecord: The audio session is allocated before calling getInput() into
audio policy serice so that the session is known before the input theead is
created and pre processings can be created on the correct session.
AudioPolicyService: default pre processing for a given input source are
loaded from audio_effects.conf file.
When an input is created, corresponding effects are created and enabled.
Change-Id: Id17119e0979b4dcf189b5c7957fec30dc3478790
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| db262e6cf072a0ebbf2b67aceb37ce9d5ba87b08 |
|
23-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix warnings for uninitialized local variables"
|
| 028ab996e50a3a6a5c6dde88f9b2cc5d2f354654 |
|
23-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Fix warnings for uninitialized local variables
Change-Id: Ic9b03b0fd215444e76c7b7bebb385f7831c557e0
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| c310dcbed66026fa548f9ec0821ec15f8c6a080d |
|
17-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Added audio_bytes_per_sample() helper function
Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 993fccec5bd0355835f0457064a28c2c13eecc60 |
|
02-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary level of priority indirection
Change-Id: I942d43973c20a7ace8b0d3f78b4da97e45e996c6
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 5439223b5633679edcc255651e7062046004d75d |
|
25-May-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Use channel mask instead of channel count for track creation
Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 34bb419e5946ab28112e9e27a4d1b3928d31e0e2 |
|
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 24fc2fb1c541e954b83fd31ea9f786a5e9b45501 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| ae29b7632ecf2068698c0d121cff284dcc82f4ec |
|
29-Mar-2011 |
Eric Laurent <elaurent@google.com> |
New fix for issue 4111672: control block flags
The first fix (commit 913af0b4) is problematic because it makes threads
in mediaserver process block on the cblk mutex. This is not permitted
as it can cause audio to skip or worse have a malicious application
prevent all audio playback by keeping the mutex locked.
The fix consists in using atomic operations when modifying the control
block flags.
Also fixed audio_track_cblk_t::framesReady() so that it doesn't block
when called from AudioFlinger (only applies when a loop is active).
Change-Id: Ibf0abb562ced3e9f64118afdd5036854bb959428
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 913af0b48fbc9a1e4db8411c763b339f2b09c954 |
|
17-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4111672: AudioTrack control block flags
Make sure that all read/modify/write operations on the AudioTrack
and AudioRecord control block flags field are protected by the
control block's mutex.
Also fix potential infinite loop in AudioTrack::write() if the
written size is not a multiple of frame size.
Change-Id: Ib3d557eb45dcc3abeb32c9aa56058e2873afee27
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 421ddc014b31478ba0ef115e098d8780a35af331 |
|
07-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3439872: video chat and bluetooth SCO
This change fixes the stability problems experienced when using
a bluetooth headset supporting both A2DP and SCO. Problems occur
when starting the video chat at which time the A2DP output is being
stopped to start SCO. At that time, active AudioTracks are invalidated
by AudioFlinger so that a new AudioTrack binder interface can be
recreated by the client process on the new mixer thread with correct parameters.
The problem was that the process to restore the binder interface was not
protected against concurrent requests which caused 2 binder interfaces
to be created sometimes. This could lead to permanent client deadlock
if one of the client threads was waiting for a condition of the first
created binder interface while the second one was created (as the AudioFlinger
would only signal conditions on the last one created).
This concurrent request situation is more likely to happen when a client
uses the JAVA AudioTrack as the JNI implementation uses simultaneously the
native AudioTrack callback and write push mechanisms. By doing so, the code
that checks if the binder interface should be restored (in obtainBuffer()) is
much more likely to be called concurrently from two different threads.
The fix consists in protecting the critical binder interface restore phase
with a flag in the AudioTrack control block. The first thread acting upon the binder
interface restore request will raise the flag and the second thread will just wait for
a condition to be signaled when the restore process is complete.
Also protected all accesses to the AudioTrack control block by a mutex to prevent
access while the track is being destroyed and restored. If a mutex cannot be held
(e.g because we call a callback function), acquire a strong reference on the IAudioTrack
to prevent its destruction while the cblk is being accessed.
Modified AudioTrack JNI to use GetByteArrayElements() instead of
GetPrimitiveArrayCritical() when writing audio buffers. Entering a critical section would
cause the JNI to abort if a mediaserver crash occurs during a write due to the AudioSystem
callback being called during the critical section when media server process restarts.
Anyway with current JNI implementation, either versions do not copy data most of the times
and the criticial version does not guaranty no data copy.
The same modifications have been made to AudioRecord.
Change-Id: Idc5aa711a04c3eee180cdd03f44fe17f3c4dcb52
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| f3d6dd0782e3135885efdc26cb86cf0bf811749f |
|
18-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3157123.
Use a Mutex wherever atomic operations were used in AudioTrack,
AudioRecord, AudioFlinger and AudioEffect classes.
Change-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 97d61f751899d3d68f53fb14d0025766bf1be3f7 |
|
22-Jun-2010 |
Chia-chi Yeh <chiachi@android.com> |
media: add AudioRecord::getMinFrameCount().
Change-Id: I952071ab10aa49aa96b727d157b68470d69fff3d
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 65b65459e6ac59f8a257009df8014467ae0838ee |
|
02-Jun-2010 |
Eric Laurent <elaurent@google.com> |
Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications.
First drop of audio framework modifications for audio effects support.
- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()
- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.
- IAudioTrack:
Added method to attach auxiliary effect.
- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.
Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.
-AudioMixer
Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel
Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| eb8f850d0b7e53956e917fd9645f808c1a09bc88 |
|
14-May-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| ba8811f5528404527b0cbad584a836f0b1807d26 |
|
03-Mar-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2428563: Camera rendered inoperable by voice call interruption.
The problem is that AudioRecord never exits read() when a timeout occurs while trying
to get new PCM data from audio hardware input buffer: it just keeps waiting and retrying until stop() is called.
In the same time, opencore AndroidAudioInput::audin_thread_func() loop cannot be exited when stuck
in AudioRecord::read() because the iExitAudioThread flag can only be sampled when AudioRecord::read()
returns. We remain stuck with the audio input thread running.
The fix consists in modifying AudioRecord behavior in case of timeout when getting new PCM samples.
We now wait only one timeout period and try to restart audio record, in case the problem is due to a media_server
process crash. If this fails, we exit read() with a number of bytes read equals to 0 so that
AndroidAudioInput::audin_thread_func() loop can exit.
Also modified Audioflinger::RecordThread() loop so that we attempt to recover from HAL read errors.
In case of read error, the input stream is forced to standby so that next read attempt does a
reconfiguration and restart of the audio input device.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 47d0a9264fa5297db6333697ad750e6bc06822aa |
|
26-Feb-2010 |
Eric Laurent <elaurent@google.com> |
Issue 2071329: audio track is shorter than video track for video capture on sholes
Add API to retrieve number of frames dropped by audio input kernel driver.
Submitted on behalf of Masaki Sato <masaki.sato@motorola.com>
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 6faf7893b6307a3295993380d61af49f2cda965c |
|
26-Jan-2010 |
Mathias Agopian <mathias@google.com> |
Simplify the MemoryDealer implementation
At some point the implementation became complicated because of
SurfaceFlinger's special needs, since we are now relying on gralloc
we can go back to much simpler MemoryDealer.
Removed HeapInterface and AllocatorInterface, since those don't need
to be paramterized anymore. Merged SimpleMemory and Allocation.
Made SimplisticAllocator non virtual.
Removed MemoryDealer flags (READ_ONLY, PAGE_ALIGNED)
Removed a lot of unneeded code.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 9cc489a2196e449dbe090ad9bdbc5cedd0b8334f |
|
05-Dec-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2304669: VoiceIME: starting and canceling voice IME yields persistent "error 8" state on future attempts and breaks voice search.
Fixed AudioFlinger::openInput() broken in change ddb78e7753be03937ad57ce7c3c842c52bdad65e
so that an invalid IO handle (0) is returned in case of failure.
Applied the same correction to openOutput().
Modified RecordThread start procedure so that a failure occuring during the first read from audio input stream is detected and causes
the record start to fail.
Modified RecordThread stop procedure to make sure that audio input stream fd is closed before we exit the stop function.
Fixed AudioRecord JAVA and JNI implementation to take status of native AudioRecord::start() into account
and not change mRecordingState to RECORDSTATE_RECORDING if start fails.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 49f02be9d7fafb8841f75394a90f1409c6f82c1f |
|
19-Nov-2009 |
Eric Laurent <elaurent@google.com> |
Issue 2265163: Audio still reported routed through earpiece on sholes
This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.
The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.
The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| bda7469d9b1ec6d9c9d6da40ddf64dc39ff271a9 |
|
04-Nov-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2203561: Sholes: audio playing out of earpiece.
Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.
Same modifications for AudioRecord.
Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| a553c25b33c99b345cf1c8688f8df0ed8df14e5a |
|
17-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1795088 Improve audio routing code
Initial commit for review.
Integrated comments after patch set 1 review.
Fixed lockup in AudioFlinger::ThreadBase::exit()
Fixed lockup when playing tone with AudioPlocyService startTone()
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| fcc5be99e0497f25a12e77622c27b4de30e45b3d |
|
07-Jul-2009 |
Eric Laurent <elaurent@google.com> |
am 88e209dc: Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
Merge commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c'
* commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c':
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
|
| 88e209dcf8c2ebddda5c272f46d1bd5478bc639c |
|
07-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| c5d0343b677d5ddc2d69fd15b81b084a40ab12ac |
|
26-May-2009 |
Android (Google) Code Review <android-gerrit@google.com> |
am de8268d6: Merge change 2331 into donut
Merge commit 'de8268d6d1cd168510c490b17e93154d2eab767c'
* commit 'de8268d6d1cd168510c490b17e93154d2eab767c':
Fix issue 1846343 - part 1
|
| 4bc035a65cac177be9294e69f110497e3b6e34e6 |
|
22-May-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1846343 - part 1
This change is the first part of a fix for issue 1846343, :
- Added new enum values for input sources in AudioRecord and MediaRecorder for voice uplink, downlink and uplink+downlink sources.
- renamed streamType to inputSource in all native functions handling audio record.
A second change is required in opencore author driver and android audio input to completely fix the issue.
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 0795272aa226f4e965968a03daddc53ce30b7cda |
|
20-May-2009 |
Mathias Agopian <mathias@google.com> |
move libbinder's header files under includes/binder
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| ef02827d4c3b9c0601eddc9c348fc2ea866420a2 |
|
21-Apr-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 1745312: Various cleanups in media framework
AudioTrack, AudioRecord:
- remove useless mAudioFlinger member of AudioTrack and AudioRecord.
- signal cblk.cv condition in stop() method to speed up stop completion.
- extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily
AudioFlinger:
- remove some warnings in AudioFlinger.cpp.
- remove function AudioFlinger::MixerThread::removetrack_l() as its content is never executed.
- remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().
- Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.
AudioSystem.java:
- correct typo in comment
IAudioflinger, IAudioFlingerClient:
- make AudioFlinger binder interfaces used for callbacks ONEWAY.
AudioHardwareInterface:
- correct routeStrings[] table in AudioHardwareInteface.cpp
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 272beb63834e213ee35a83741a5e2ae1279999d5 |
|
25-Mar-2009 |
Eric Laurent <> |
Automated import from //branches/donutburger/...@142065,142065
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 78c131473165830c8d042a0f445ca858aaa7a4b1 |
|
25-Mar-2009 |
Jean-Michel Trivi <> |
Automated import from //branches/donutburger/...@141200,141200
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 9066cfe9886ac131c34d59ed0e2d287b0e3c0087 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| d83a98f4ce9cfa908f5c54bbd70f03eec07e7553 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 076357b8567458d4b6dfdcf839ef751634cd2bfb |
|
03-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@132589
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 3dec7d563a2f3e1eb967ce2054a00b6620e3558c |
|
03-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@137055
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 15ab3eae2ec3d73b3e8aa60b33ae41445bf83f4b |
|
20-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@132569
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| da996f390e17e16f2dfa60e972e7ebc4f868f37e |
|
13-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@131421
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| d24b8183b93e781080b2c16c487e60d51c12da31 |
|
11-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@130745
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 22f7dfd23490a3de2f21ff96949ba47003aac8f8 |
|
20-Jan-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@127101
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| f013e1afd1e68af5e3b868c26a653bbfb39538f8 |
|
18-Dec-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Code drop from //branches/cupcake/...@124589
/frameworks/base/media/libmedia/AudioRecord.cpp
|
| 54b6cfa9a9e5b861a9930af873580d6dc20f773c |
|
21-Oct-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Initial Contribution
/frameworks/base/media/libmedia/AudioRecord.cpp
|